[..] |
ASTERISK-06000: [patch] pbx_spool.c: leaks memory on DelayedRetry |
ASTERISK-06001: [patch] When using ODBCSTORAGE, running as non-root, can't write to temp messages |
ASTERISK-06002: Deadlock for app_queue when sending it high volume of calls. |
ASTERISK-06003: [patch] use ast_skip_blanks instead of rewriting it inline |
ASTERISK-06004: [patch] simplify code in file.c |
ASTERISK-06005: [patch] constify manager_event arguments |
ASTERISK-06006: [patch] app_playbac.c code simplifications |
ASTERISK-06007: [patch] enable ast_skip_blanks on const char * |
ASTERISK-06008: [patch] mnor code tweaks |
ASTERISK-06009: [patch] allow compilation if ZT_EVENT_RINGBEGIN is undefined |
ASTERISK-06010: MWI (SIP NOTIFY) not sent when using Realtime Voicemail |
ASTERISK-06011: Mantis Problems |
ASTERISK-06012: dblock blocking forever |
ASTERISK-06013: asterisk does not properly drop root privileges |
ASTERISK-06014: [patch] extra ast_mutex_unlock() in manager.c::ast_manager_register2() |
ASTERISK-06015: misplaced ast_moh_stop in svn7865 ? |
ASTERISK-06016: [patch] chan_sip.c::realtime_update_peer may update with uninitialized variables |
ASTERISK-06017: error with modconf when installing zaptel 1.2.1 |
ASTERISK-06018: Zaptel PRI lines dead after incoming AGI call terminates early |
ASTERISK-06019: [patch] Segfault in pbx_builtin_setvar_helper. Looks like the channel is being closed down underneath us |
ASTERISK-06020: [patch] make uninstall/uninstall-all |
ASTERISK-06021: [patch] Set presentation to allowed whenever we set a callerid component |
ASTERISK-06022: [patch] ast_strlen_zero janitor project - NULL check found in func_md5.c |
ASTERISK-06023: data type for 'recording' parameter should be SQL_LONGVARBINARY, not SQL_BINARY |
ASTERISK-06024: Problems caused by MySQL realtime driver |
ASTERISK-06025: chan_sip schedules destruction of subscription twice (causes crash) |
ASTERISK-06026: overriding callerid in sip.conf does not work |
ASTERISK-06027: missing mutex randomlock in svn7920 |
ASTERISK-06028: [patch] use argument macro in pbx.c |
ASTERISK-06029: [patch] use argument macro in func_md5.c |
ASTERISK-06030: [patch] use argument macro in func_math.c |
ASTERISK-06031: Makefile generates different libpri.so.1.0 to the exported libpri.so.1 |
ASTERISK-06032: Makefile generates different libtonezone.so.1.0 to the exported libtonezone.so.1 |
ASTERISK-06033: [patch] extra statement causing memory leak on unload in chan_sip.c |
ASTERISK-06034: [patch] Spelling error in channel.c |
ASTERISK-06035: [patch] VMAuthenticate does not recognize provided mailbox |
ASTERISK-06036: [patch] CDR not ended before 'h' extension |
ASTERISK-06037: codecpriority=caller does not seem to work |
ASTERISK-06038: chan_phone deadlocks |
ASTERISK-06039: [patch] app_queue has a deadlock when weight is used |
ASTERISK-06040: [patch] local variable used when it are not valid |
ASTERISK-06041: [patch] use argument macro in func_cut.c |
ASTERISK-06042: [patch] use argument macro in func_cut.c |
ASTERISK-06043: [patch] use argument macro in func_cut.c |
ASTERISK-06044: Extra unlocks in app_queue weights |
ASTERISK-06045: [patch] wrong return from build_peer() |
ASTERISK-06046: [patch] use argument macro in func_cdr.c |
ASTERISK-06047: [patch] use argument macro in func_cdr.c |
ASTERISK-06048: [patch] Converting /apps to use *alloc() wrappers - 1 |
ASTERISK-06049: Asterisk doesn't detect answer for some numbers |
ASTERISK-06050: [patch] New App Saynumber, for generic way of easy internationalization |
ASTERISK-06051: Asterisk Core Dumps |
ASTERISK-06052: [patch] API Manager report a unique id value, in place of time, in AgentLogoff event |
ASTERISK-06053: [patch] Converting /apps to use *alloc() wrappers - 2 |
ASTERISK-06054: app_voicemail realtime uses a SQL92 reserved word (delete) |
ASTERISK-06055: [test-this-branch] added support for "meetme list <confno> concise" to easily script parsing |
ASTERISK-06056: [patch] app_morsecode |
ASTERISK-06057: Documentation for new auto-play option (report #0006090) for VoiceMailMain |
ASTERISK-06058: RemoteVoiceMailMain |
ASTERISK-06059: [patch] decouple struct localuser from private application variables |
ASTERISK-06060: vmodem.h not used anymore ? |
ASTERISK-06061: SIP queue members are flagged 'unavailable' while definetely available... |
ASTERISK-06062: is createlink still use in agent.conf? |
ASTERISK-06063: [patch] missing ast_unloc() in pbx.c::__ast_context_destroy() |
ASTERISK-06064: [patch] use argument macro in app_voicemail.c |
ASTERISK-06065: [patch] use argument macro in app_dictate.c |
ASTERISK-06066: compile problem on freebsd 4.11 - missing header <stdint.h> |
ASTERISK-06067: [patch] Converting /apps to use *alloc() wrappers - 3 |
ASTERISK-06068: [patch] Converting /apps to use *alloc() wrappers - 4 |
ASTERISK-06069: [patch] spell check in chan_agent.c |
ASTERISK-06070: [patch] macro and other change in chan_agent.c |
ASTERISK-06071: [patch] Fix for answer & hangup on polarity reversal when both [answer/hangup]onpolarityswitch=yes |
ASTERISK-06072: GotoIfTime function does not handle the [weekdays] parameter properly |
ASTERISK-06073: [patch] code clean and minor change in chan_agent.c |
ASTERISK-06074: asterisk crashing in ast_cdr_alloc on AMD64 platform. |
ASTERISK-06075: asterisk 1.2.1, asterisk-sounds-1.2.1 duplicated files |
ASTERISK-06076: [patch] option missing in agent.conf |
ASTERISK-06077: [patch] another plase to use argument macro in chan_agent.c |
ASTERISK-06078: [patch] another plase to use argument macro in chan_agent.c |
ASTERISK-06079: [patch][post 1.4] adjust volume of voicemail sent by email |
ASTERISK-06080: [patch] macro and other change in func_string.c |
ASTERISK-06081: [patch] ascii time format in queue_log |
ASTERISK-06082: [branch] Errors in support for SIP strict routing |
ASTERISK-06083: [patch] app_uservent does not parse arguments correctly |
ASTERISK-06084: [patch] Add manager actions DBDel and DBDeltree |
ASTERISK-06085: [discussion] Need a way to deprecate manager APIs |
ASTERISK-06086: [patch] macro in manager.c, func_rand.c, func_db.c |
ASTERISK-06087: [patch] array function in func_string don't work |
ASTERISK-06088: MixMonitor crash? |
ASTERISK-06089: [patch] macro in manager.c, func_rand.c, func_db.c |
ASTERISK-06090: [patch] macro in manager.c, func_rand.c, func_db.c |
ASTERISK-06091: [patch] spell check in some files |
ASTERISK-06092: [patch] redundant cflags in subdir makefiles |
ASTERISK-06093: [patch][post 1.4] Add possibility to set Zap channels into hdlc mode |
ASTERISK-06094: [patch] use argument macro in func_string.c |
ASTERISK-06095: add an additional return value to app_record.c |
ASTERISK-06096: Race condition when do threeway |
ASTERISK-06097: PRI calls to just ONE number dropped with NOANSWER between pstn and Alcatel |
ASTERISK-06098: [patch] QUOTE function |
ASTERISK-06099: assorted pbx.c issues (some bugs, some don't know) |
ASTERISK-06100: D-channel giving up when allready done |
ASTERISK-06101: [patch] Cleaning up apps/app_sql_postgres.c |
ASTERISK-06102: Added STRPTIME to func_strings.c to complement STRFTIME |
ASTERISK-06103: [patch] "queue_log = no" not work |
ASTERISK-06104: [branch] app_dial: Privacy option 2 returns dial-status ANSWER / option_priority_jumping not respected |
ASTERISK-06105: Problems in Voicemail |
ASTERISK-06106: with 'o' flag on, your own voice is played back to you in the conference |
ASTERISK-06107: [patch] macro and other change in app_queue.c |
ASTERISK-06108: chan_sip.c deadlock? |
ASTERISK-06109: Asterisk crashed making calls using IAX2 |
ASTERISK-06110: [patch] constification of some fields in pbx.c |
ASTERISK-06111: [patch] fix for memory overwrite in pbx.c::substring() |
ASTERISK-06112: [patch] improve documentation of cli.h and remove redundant 'extern' |
ASTERISK-06113: chanspy to sipura 1001 |
ASTERISK-06114: [patch] simplify show uptime handler |
ASTERISK-06115: [patch] Converting /apps to use *alloc() wrappers - 5 |
ASTERISK-06116: Asterisk crashes when a call is blind-transfered to Parking |
ASTERISK-06117: [patch] Converting /codecs to use *alloc() wrappers |
ASTERISK-06118: [patch] chan_sip sends malformed voicemail notification if fromdomain is not set |
ASTERISK-06119: option for caller to hear call progress in app_queue |
ASTERISK-06120: [branch] rewritten ast_extension_{match|core} |
ASTERISK-06121: SIPAddHeader add only 1 header |
ASTERISK-06122: queue timeout is ignored |
ASTERISK-06123: asterisk multiple dead processes |
ASTERISK-06124: SIP PROTOCOL VIOLATION: Route information ignored when sending BYE requests. |
ASTERISK-06125: MOH not working for calls placed to agent |
ASTERISK-06126: [patch] constification of *_complete_*() functions |
ASTERISK-06127: ztdummy no longer builds against kernel 2.6.16-rc1 |
ASTERISK-06128: wcusb will not build against 2.6.16-rc1 |
ASTERISK-06129: Abondoned Call counter wrongly incremented for Queue calls |
ASTERISK-06130: [patch] app_milliwatt cause segfault on FreeBSD 5.x, 6.0 |
ASTERISK-06131: README-1.2.2 and README-1.2.2-netsec are identical on FTP site |
ASTERISK-06132: Specify the right Rev for udptl.c |
ASTERISK-06133: The NewChannelEvent: State: Ringing is sent to the call manager ONLY if the phone is directly connected to the Asterisk PBX. |
ASTERISK-06134: GSM codec doesn't compile on ia64 |
ASTERISK-06135: small app_morsecode fixes |
ASTERISK-06136: [patch] No distinctive ring detection on TDM with cidstart=polarity |
ASTERISK-06137: RedHat init scripts should not enable Asterisk service by default |
ASTERISK-06138: Need to show Agent paused for show agents CLI command |
ASTERISK-06139: [patch] Converting /res to use *alloc() wrappers |
ASTERISK-06140: [patch] Codec formatting fixes and doxygen updates |
ASTERISK-06141: chan_agent & persistentagents=yes |
ASTERISK-06142: [patch] list process error in app_queue.c |
ASTERISK-06143: remote MWI |
ASTERISK-06144: [patch] added park command to manager interface |
ASTERISK-06145: [patch] iax2 registry does not work correctly with a dynamic dns server |
ASTERISK-06146: [patch] parked calls can't come completely out of the parking lot! |
ASTERISK-06147: Bug: Use uninicialized variable timer_t1 |
ASTERISK-06148: Bridged Sipura ATA to SIP provider call generates "503 Server Error" in a loop (Asterisk 1.2) |
ASTERISK-06149: [patch] Fixing a memory leak in res/res_odbc.c |
ASTERISK-06150: getting errors from frame.c |
ASTERISK-06151: weird crash in ast_shrink_phone_number |
ASTERISK-06152: [patch] use list macro in app_queue.c |
ASTERISK-06153: when registering to a dynamic host peer, the hostname will not be re-resolved. |
ASTERISK-06154: SRV DNS lookup does not support CNAME records |
ASTERISK-06155: [patch] disable ringing on busy members that support AST_DEVICE_INUSE |
ASTERISK-06156: New option n / Disable the conf-onlyperson message |
ASTERISK-06157: [patch] new function ast_get_time_t() |
ASTERISK-06158: No "SIPAddHeader" function |
ASTERISK-06159: messages |
ASTERISK-06160: Crash asterisk. |
ASTERISK-06161: [patch] segfault in channel.c::copy_data_from_queue - copies too many bytes |
ASTERISK-06162: [patch] implementation of SHA-1 in * |
ASTERISK-06163: build for func_odbc broken on FreeBSD |
ASTERISK-06164: incoming channel's cid presentation not passed to outgoing channel |
ASTERISK-06165: calls time out before timeout |
ASTERISK-06166: [request) Support of "SIP/2.0 300 Multiple choice" response to REGISTER request |
ASTERISK-06167: [patch] use list macro in app_meetme.c |
ASTERISK-06168: [patch] use list macro in app_voicemail.c |
ASTERISK-06169: [patch] use list macro in asterisk.c |
ASTERISK-06170: [patch] use list macro in astman.c |
ASTERISK-06171: [patch] CDR billsec tweak |
ASTERISK-06172: Given Vars into queue, have them available when leaving the queue |
ASTERISK-06173: * has lots of warnings when get rfc2833 dtmf from HuaweiSoftX3000 softswitch |
ASTERISK-06174: [patch] Asterisk ignores ACLs and umask for most file creations |
ASTERISK-06175: [patch] Realtime call control |
ASTERISK-06176: PGconn *conn and PGresult *result should be static |
ASTERISK-06177: Impossible to read voicemessages from database using ODBC. |
ASTERISK-06178: [patch] use list macro in channel.c |
ASTERISK-06179: rfc2833 dtmf is unrecognizable with 1.2.1 |
ASTERISK-06180: [patch] One Touch Parking as built-in feature |
ASTERISK-06181: Dropped calls when Operator transfers |
ASTERISK-06182: [patch] use list macro in chan_agent.c |
ASTERISK-06183: Bug in retrieve_file for ODBC. open method always return -1. |
ASTERISK-06184: ael reload does not print past 370 lines |
ASTERISK-06185: Make ${SIP_CODEC} or equivalent readable |
ASTERISK-06186: Dialstatus Returns Incorrect Results When Peer Is Unrechable |
ASTERISK-06187: Make Say*() backgroundable |
ASTERISK-06188: [patch] send/recv RDNIS information element over IAX channels |
ASTERISK-06189: [patch] No audio on bridges from jan 25 2006 |
ASTERISK-06190: [patch] use list macro in chan_feature.c |
ASTERISK-06191: asterisk -rx fails to return to command line |
ASTERISK-06192: [patch] make possible to define prefix for automon saved files |
ASTERISK-06193: Bus error on Sparc in socket_read at chan_iax2.c:5280 on asterisk 1.2.1 |
ASTERISK-06194: PATCH: Set IP TOS separately for SIP packets, RTP audio packets, and RTP video packets |
ASTERISK-06195: [patch] L option of Dial does not work properly |
ASTERISK-06196: [patch] L option of Dial does not work properly |
ASTERISK-06197: [patch] L option of Dial does not work properly |
ASTERISK-06198: [patch] L option of Dial does not work properly |
ASTERISK-06199: [patch] L option of Dial does not work properly |
ASTERISK-06200: [patch] use list macro in cli.c |
ASTERISK-06201: REGISTER without Contact: and Expires: fails |
ASTERISK-06202: CPU load increasing when a SIP client using iLBC is left alone in a MeetMe conference |
ASTERISK-06203: Unreliable inbound DTMF signalling on E&M trunks. |
ASTERISK-06204: stun client support |
ASTERISK-06205: [patch] Converting /(root) to use *alloc() wrappers - 1 |
ASTERISK-06206: Calling trough Adress Plus not working anymore |
ASTERISK-06207: Using 'p' option with MeetMe crashes Asterisk when caller presses # |
ASTERISK-06208: Failed to run ResetCDR or ForkCDR when using cdr_mysql with uniqueid enabled |
ASTERISK-06209: Change $g_bug_link_tag configuration for mantis |
ASTERISK-06210: ENUMLOOKUP count return is not as documented |
ASTERISK-06211: Unable to start after making and installing new SVN Head |
ASTERISK-06212: The output of the 'iax2 show registry'-command is wrong |
ASTERISK-06213: [patch] cleanup of file.c |
ASTERISK-06214: Realtime lookup failure causes crash? |
ASTERISK-06215: When hitting # for next channel, the # is passed on to the channel listened on |
ASTERISK-06216: [patch] deprecate CHECK_MD5() |
ASTERISK-06217: Fast AGI HANGUP fails with result=-1 |
ASTERISK-06218: MeetMe transcoding problem |
ASTERISK-06219: [patch] add queue_log_name to logger.conf and remove hardcoded queue_log |
ASTERISK-06220: [patch] Converting /(root) to use *alloc() wrappers - 2 |
ASTERISK-06221: [patch] Crash on a CLI originate command with a missing channel data |
ASTERISK-06222: [patch] Convert cdr_addon_mysql to use custom cdr format |
ASTERISK-06223: h.323 native bridge |
ASTERISK-06224: Typo in warning message |
ASTERISK-06225: [patch] use list macro in chan_local.c |
ASTERISK-06226: [patch] assorted chan_oss fixes and one improvement |
ASTERISK-06227: [patch] autoservice.c remove unnecessary include, variable, etc. |
ASTERISK-06228: [patch] MWI Subsription not working on AVM Boxes |
ASTERISK-06229: MOH in "files" mode writes in wrong formats at certain times |
ASTERISK-06230: SIP REGISTER problems with Adit 3104 |
ASTERISK-06231: [patch] AccountCode not available when called transferred (SIP REFER) |
ASTERISK-06232: After an unkonwn period of time asterisk starts refusing IAX inbound calls |
ASTERISK-06233: [patch] Converting /(root) to use *alloc() wrappers - 3 |
ASTERISK-06234: Reproduceable one-way audio problem with bridged iax2 calls that have passed DTMF |
ASTERISK-06235: [patch] chan_zap does not compile with WITHOUT_PRI |
ASTERISK-06236: Hangup event occurs when the agent answers the call - [queue related problem] |
ASTERISK-06237: Calling any Agi before Dial breaks timeout |
ASTERISK-06238: Ability to rewind the musiconhold |
ASTERISK-06239: Background does not return until sound file has finished playing. |
ASTERISK-06240: crush asterisk |
ASTERISK-06241: [patch] Display an error from chan_zap.c if ZT_SIG_HARDHDLC isn't defined |
ASTERISK-06242: res_config_mysql.so crashes when using IAX. |
ASTERISK-06243: [patch] dereference after free in file.c::ast_closestream() |
ASTERISK-06244: Grandstream Devices don't hangup with Asterisk |
ASTERISK-06245: IAX2 BLINDTRANSFER channel variable support |
ASTERISK-06246: alternative timer sources |
ASTERISK-06247: Incoming SIP calls to empty usernames do not trigger 's' extension rules |
ASTERISK-06248: [patch] fix chan_zap compilation with old zaptel and !ZAPATA_PRI |
ASTERISK-06249: can't get status device |
ASTERISK-06250: patch close some bugg (avoid dead lock with s.transfer && bad cdr ) |
ASTERISK-06251: dst field overwritten |
ASTERISK-06252: Leading DTMF lost due to lack of delay on Zapata/TDM400P/X100M |
ASTERISK-06253: app_dial connects but no voice |
ASTERISK-06254: Please backport to 1.2: Loosen voicemail file permissions to allow group access |
ASTERISK-06255: [patch] make amaflags and dispoisition available as integer value |
ASTERISK-06256: [patch][post 1.4] make fields configurable |
ASTERISK-06257: app_sql_postgres fails to build (Asterisk SVN 1.2.4) |
ASTERISK-06258: asterisk opens but cannot play back speex encoded files |
ASTERISK-06259: [patch] alaw calls unsupported (Asked to get a channel of unsupported format '8') |
ASTERISK-06260: Meetme does not release channels when using 3rd party driver(CHAN_SCCP2) |
ASTERISK-06261: Unreachable vars when agent is called back. |
ASTERISK-06262: MixMonitor causes segfault on Free BSD 5.4 |
ASTERISK-06263: Crash in chan_sip |
ASTERISK-06264: SIP Hold on Avaya 4610 kills iax leg of call |
ASTERISK-06265: CDR variables doesn't have the real values i.e CDR(duration), CDR(billsec) and CDR(end) |
ASTERISK-06266: SIP transfers of queued calls doesn't make agent available neighter makes new agent busy |
ASTERISK-06267: Unable to monitor a call in app_queue. |
ASTERISK-06268: When SIP is bound to 0.0.0.0 it answers only on the first interface |
ASTERISK-06269: [patch] queue timeout unexpected behaviour |
ASTERISK-06270: G729 codec crashed Asterisk and safe_asterisk |
ASTERISK-06271: [patch] function returns incorrect value |
ASTERISK-06272: [patch] Converting /(root) to use *alloc() wrappers - 4 |
ASTERISK-06273: Cdr_tds does not compile at all |
ASTERISK-06274: Error message needed for brand new asterisk installation without any samples |
ASTERISK-06275: Configurable instruction sound per context (like directoryintro=) |
ASTERISK-06276: SNMP [Sub]Agent for Asterisk |
ASTERISK-06277: Tandem Asterisk fails |
ASTERISK-06278: [patch] KEYPADHASH |
ASTERISK-06279: peercontext is not updated on reload |
ASTERISK-06280: Mantis bug in [print] |
ASTERISK-06281: AgentMonitorOutgoing when bridged |
ASTERISK-06282: chan_iax2 hangs intermittently |
ASTERISK-06283: [patch] Compilation problem with chan_h323 and gcc 3.3.2 |
ASTERISK-06284: Proxy Forking |
ASTERISK-06285: Error if pbx_functions.so is not loaded |
ASTERISK-06286: MWI notify on Polycom IP501 with multiple registrations |
ASTERISK-06287: MIXMONITOR_FILENAME is not available on SIP -> ZAP calls for script processing. |
ASTERISK-06288: [patch] compile problem - missing prototype for snprintf |
ASTERISK-06289: [patch] Makefile problem in channels/ directory |
ASTERISK-06290: app_page compile error due to function signature change in pbx.c |
ASTERISK-06291: [patch] uninitialized vairable in function_sippeer() |
ASTERISK-06292: LOCAL_USER_ADD and LOCAL_USER_ACF_ADD are now the same |
ASTERISK-06293: [patch] chan_h323 memory leak |
ASTERISK-06294: [patch] ilbc library not rebuilt causing module load failure. |
ASTERISK-06295: PRI Channels Blocking and unavailable with PRI "Call" flag set true. |
ASTERISK-06296: Makefile Does Not Work on Tru64 UNIX 5.4 |
ASTERISK-06297: Make fails on HP-UX 11 |
ASTERISK-06298: The Busy (or the Congestion) command can switch the channel in a zombie state |
ASTERISK-06299: AGI Does not complete running before terminating. |
ASTERISK-06300: Compile error, segmentation fault |
ASTERISK-06301: dest variable in ENUMLOOKUP/func_enum.c is too small |
ASTERISK-06302: [patch] extensions.conf.sample has incorrect syntax for GotoIf in macro-page context. |
ASTERISK-06303: Make fails on OpenBSD 3.6 |
ASTERISK-06304: Addons - OH323 does not link on OS/X |
ASTERISK-06305: Asterisk crashes several times a day |
ASTERISK-06306: duplicated IAX peers result in crash on startup |
ASTERISK-06307: No Audio When bridging to vpb |
ASTERISK-06308: Wrong greeting replayed when re-recording a message (review=yes) |
ASTERISK-06309: not dumping core when run as non-root |
ASTERISK-06310: [patch] app_meetme does not always turn off MOH when exiting |
ASTERISK-06311: Zaptel fails to compile |
ASTERISK-06312: Directory Application doesn't work if using realtime config - 0002475 |
ASTERISK-06313: RINGING event is not send when I place calls to Cisco GW. |
ASTERISK-06314: App_queue crashes asterisk when using chan_sccp |
ASTERISK-06315: [patch] app_dial reports CHANUNAVAIL on telco intercept messages |
ASTERISK-06316: [patch] new agi function, NiceAGI |
ASTERISK-06317: Unable to assign channel variable to macro agument inside a macro |
ASTERISK-06318: [patch] wrong code in pbx.c SVN10088 |
ASTERISK-06319: [patch] STANDARD_LOCAL_USER removal |
ASTERISK-06320: SIP to SIP calls randomly result in one-way calls |
ASTERISK-06321: [patch] Asterisk binary linked with openssl |
ASTERISK-06322: [patch] backtrace log must be LOG_DEBUG, not LOG_WARNING |
ASTERISK-06323: [patch] KEYPADHASH don't work with '*' and '#' |
ASTERISK-06324: Initial burst of calls is causing asterisk to have problems with the management connections |
ASTERISK-06325: Text error in README |
ASTERISK-06326: Calls from mgcp to SIP and mISDN channel got failed |
ASTERISK-06327: Channel moving goes bad |
ASTERISK-06328: [patch] Clarify 'mailbox' terminology |
ASTERISK-06329: [branch][post 1.4] Make storage of voicemail and greetings slightly more abstract |
ASTERISK-06330: [patch] xpp_usb module won't compile with kernel >= 2.6.16-rc1 |
ASTERISK-06331: [patch] Add CLI commands for viewing global variables and setting global variables |
ASTERISK-06332: Using the directory to forward a message uses wrong context when context not specified in parameter |
ASTERISK-06333: [patch] app_asyncgoto |
ASTERISK-06334: CONGESTION gallore in the afternoon |
ASTERISK-06335: asterisk crash when exiting app_meetme with native moh |
ASTERISK-06336: [patch] More Q.SIG-specific cause codes for debugging |
ASTERISK-06337: [branch][post 1.4] Initial ast_vm_message abstraction |
ASTERISK-06338: ENUMLOOKUP failing on all methods |
ASTERISK-06339: Extension matching with the . wildcard does not obey digit timeout |
ASTERISK-06340: [patch] extra ';' in AST_LIST_HEAD_STATIC and AST_LIST_HEAD_NOLOCK_STATIC |
ASTERISK-06341: [patch] uninitialized field in app_exec() |
ASTERISK-06342: [PATCH] bitwise and used instead of modulus in indexing history in jitterbuffer history recalculation |
ASTERISK-06343: [patch] Converting /(root) to use *alloc() wrappers - 5 |
ASTERISK-06344: add custom fields to cdr_mysql |
ASTERISK-06345: [patch][post 1.4] add context filter in "zap show channels" |
ASTERISK-06346: Compilation problem with chan_h323 and gcc 4.0.1 |
ASTERISK-06347: [patch] Record bridge channel unique id in "CONNECT" queue_log entry |
ASTERISK-06348: [patch] Called party wrongly transfered to priority+1 when G() option used with Dial |
ASTERISK-06349: [patch] Disable log / verbose output to remote consoles |
ASTERISK-06350: [patch] Failure to lock voicemail.conf leads to corruption |
ASTERISK-06351: [patch] app_meetme crashes Asterisk in rare/odd scenario where first caller hangs up during pin request. |
ASTERISK-06352: [branch] devicestate of a hinted SIP phone does not update on outbound calls if call-limit is specified. |
ASTERISK-06353: README and UPGRADE.txt update |
ASTERISK-06354: [patch] Memory leak in app.c |
ASTERISK-06355: [patch] app_amd: be consistent trough the code, documentation updates |
ASTERISK-06356: Local channels do not change to their connected sip trunk upon receiving Answer |
ASTERISK-06357: [patch] Allow libpri to handle multiple D-channels on single signalling link |
ASTERISK-06358: [patch] useadsi is not always initialized |
ASTERISK-06359: __list_prev not treated properly in list deletion in AST_LIST_TRAVERSE_SAFE_BEGIN |
ASTERISK-06360: [patch] useadsi is not always initialized |
ASTERISK-06361: Makefile:96: *** invalid syntax in conditional. Stop. |
ASTERISK-06362: [patch] trivial comments cleanup in pbx.h |
ASTERISK-06363: [patch] iLBC quality |
ASTERISK-06364: [patch] format_sln -- ignoring error returned from fseek |
ASTERISK-06365: [patch] starting to using off_t type and fseeko/ftello |
ASTERISK-06366: [patch] ast_rtp_read( rtp ) hasn't check that rtp != NULL |
ASTERISK-06367: [patch] channel.c ast_channel_make_compatible remove duplicate code |
ASTERISK-06368: App_Callback doesnt compile on Asterisk 1.2.4 |
ASTERISK-06369: [patch] MOH continues to run monmp3thread after freeing class |
ASTERISK-06370: crash on startup |
ASTERISK-06371: crash using IAX client connected to SpyChannel,IAX2 |
ASTERISK-06372: If I use the 'reload' command in asterisk, asterisk crashes with a segmentation fault. |
ASTERISK-06373: [request] Enable Monitor when the RTP is not going through the Asterisk |
ASTERISK-06374: Setup Message |
ASTERISK-06375: ast_softhangup_nolock does not set hangupcause correctly |
ASTERISK-06376: When dialling multiple Channels disposition will always be FAILED even if the Call is ANSWERED |
ASTERISK-06377: PRI Configuration Error doesn't say which span it's complaining about |
ASTERISK-06378: cdr_odbc created lots of connections to db |
ASTERISK-06379: [patch] add "pri show spans" CLI command |
ASTERISK-06380: [patch] app_trunkisavail.c - load balancing between mulitple trunks |
ASTERISK-06381: User/friend registrations timing out prematurely |
ASTERISK-06382: crash on voicemail config reload |
ASTERISK-06383: [patch] chan_sip possible deadlock |
ASTERISK-06384: [patch] Add new queue_log event |
ASTERISK-06385: [patch] Callback prototype mismatch causes crash |
ASTERISK-06386: The command "show channels" display a wrong number of calls and generate "Avoided deadlock" warnings |
ASTERISK-06387: app_skel.c does not compile |
ASTERISK-06388: Asterisk 1.2.4 fails to send request for RFC2833 |
ASTERISK-06389: SIP bridging fails with no audio unless... |
ASTERISK-06390: [patch] Add existence / permission check before running a script |
ASTERISK-06391: Makefile doesn't honor DESTDIR |
ASTERISK-06392: Makefile doesn't honor DESTDIR |
ASTERISK-06393: dynamic payload type |
ASTERISK-06394: [patch] _ast_request_and_dial() fails if the channel is immediately UP |
ASTERISK-06395: [patch] Asterisk crashes randomly when using manager to generate predictive calls with Zap and SIP |
ASTERISK-06396: S(timeout) is broken |
ASTERISK-06397: Missing symbol |
ASTERISK-06398: Asterisk Segfaulted (no idea why) |
ASTERISK-06399: Retransmission of SIP-INFO DTMF causes invalid CSeq error on Cisco 5300 |
ASTERISK-06400: [patch] Add timeout option and return result in separate variable in CURL app/func |
ASTERISK-06401: SIP_CODEC not working for 182 Session Progress / Early media |
ASTERISK-06402: Asterisk crashes when using Realtime through ODBC |
ASTERISK-06403: Codec order is not working ok |
ASTERISK-06404: The CDR report the "src" and "dst" equal in any application |
ASTERISK-06405: [patch] Define platform-specific ASTxxxDIRs in one single place in Makefile |
ASTERISK-06406: [patch] Broken MALLOC_DEBUG build |
ASTERISK-06407: [patch] Converting /(root) to use *alloc() wrappers - 6 |
ASTERISK-06408: Suggestion - Dial() parameter - add functionality of calling a macro every 'z' ms. |
ASTERISK-06409: func_curl fails to compile |
ASTERISK-06410: callwaiting=yes not functional |
ASTERISK-06411: FastAGI - HANGUP - 510 Invalid or unknown command |
ASTERISK-06412: crashed under heavy load with group_count |
ASTERISK-06413: Kernel NMI Error (possibly Zaptel Related) causes asterisk to die |
ASTERISK-06414: [patch] Add option to display channel variables in AgentCalled events |
ASTERISK-06415: Uniden UIP 200 with Asterisk V 1.2.X |
ASTERISK-06416: Congestion function can hang system by consuming all available bandwidth |
ASTERISK-06417: [patch] Converting /(root) to use *alloc() wrappers - 7 |
ASTERISK-06418: Unable to find a codec translation path from alaw to slin |
ASTERISK-06419: [patch] res_config_odbc.so not linked with odbc |
ASTERISK-06420: [patch] app_waitforsilence is broken |
ASTERISK-06421: I have the following dial path, it places the call ok but does not pass audio |
ASTERISK-06422: "sip show registry" shows an incorrect remote port when using "register => 2345:password@sip_proxy" syntax |
ASTERISK-06423: Request: Caller/Callee ability to disable music on hold |
ASTERISK-06424: [patch] A minor issue in meetme join/leave manager events |
ASTERISK-06425: [patch] Integration with ZIM-SMS (www.zim.biz) |
ASTERISK-06426: [patch] Asterisk init.d file does not allow for the -C option |
ASTERISK-06427: astvarlibdir option in asterisk.conf does not work for keys |
ASTERISK-06428: [patch] Russian locale for voicemail |
ASTERISK-06429: [patch] cli tries to print usage even if there isn't one |
ASTERISK-06430: meetme-admin menu: 2 functions not working |
ASTERISK-06431: MeetMe enter/leave sounds are still not customizable |
ASTERISK-06432: SetCallerID and SetCallerPres will be ignored if Caller Leg is a CAPI call |
ASTERISK-06433: [PATCH] 'iax show peer ...' incorrectly always shows qualify smoothing as disabled, regardless of reality. |
ASTERISK-06434: command line options do not override config defaults |
ASTERISK-06435: asterisk build on intel iMac crashes or gets into a race condition.... |
ASTERISK-06436: one-dimensional MusicOnHold |
ASTERISK-06437: [branch] [post 1.4] peermatch: Changes to users/peers, while being backwards compatible |
ASTERISK-06438: sounds/voicemail dir keeps pointing to /var/spool/asterisk/voicemail even when asterisk.conf is changed |
ASTERISK-06439: [patch] fixes for a couple of compile problems. |
ASTERISK-06440: Integer values of CDR are not corrcetly bound by ODBC driver for MySQL |
ASTERISK-06441: asterisk-1.2.x iax.conf bindport= is ignored if bindaddr= is not present |
ASTERISK-06442: VMAuthenticate's Silent Option Does not work |
ASTERISK-06443: [patch] Quotes missing for sed command in build_tools/make_svn_branch_name |
ASTERISK-06444: [patch] The command [[ in funcs/Makefile is not recognized in Solaris sh |
ASTERISK-06445: [need disclaimer] libiax2 crashes in iax2-parser.c iax_showframe() |
ASTERISK-06446: ParkAndAnnounce calls Asterisk to crash. |
ASTERISK-06447: Asterisk became unresponsive after catching an error and generating 70meg of SIP traffic directed towards SIP Media Gateway. |
ASTERISK-06448: scan-service giving an strange error |
ASTERISK-06449: Polycom 601 returns Internal Server Error 500 in response to SIP NOTIFY messages |
ASTERISK-06450: [patch] improve indication of peer status in "sip show peers" |
ASTERISK-06451: Complete system freeze caused, I believe, by chan_misdn. |
ASTERISK-06452: Queues freeze if AgentCallbackLogin is used |
ASTERISK-06453: "sip reload" puts port numbers to 5060. |
ASTERISK-06454: [patch] Res_snmp/libnetsnmp collides with unload_module |
ASTERISK-06455: Allow configurable table name in cdr_tds.c |
ASTERISK-06456: ChangeMonitor returns wrong message upon success |
ASTERISK-06457: ChangeMonitor deos not change the filenames |
ASTERISK-06458: reinvite - audio delay |
ASTERISK-06459: [patch] groups from within voicemail |
ASTERISK-06460: [branch] siptransfer: Improving the REFER support |
ASTERISK-06461: [patch] use readline instead of editline |
ASTERISK-06462: [patch] Compilation of asterisk.c on Solaris generates several warnings |
ASTERISK-06463: [patch] build_tools/make_build_h uses whoami, a command that doesn't exist in Solaris |
ASTERISK-06464: [branch] cdr_radius module |
ASTERISK-06465: Privacy Manager does not work unless caller presses '#'. |
ASTERISK-06466: Copy/paste error in chan_misdn commit r11714 |
ASTERISK-06467: [patch] Implement Called Party Identification |
ASTERISK-06468: Wrong account selected for inbound calls |
ASTERISK-06469: SIP Responds on incorrect IP |
ASTERISK-06470: [patch] Compiling editline/term.c on Solaris generates several warnings |
ASTERISK-06471: Channel mutex may be locked before ast_hangup() call in pbx.c |
ASTERISK-06472: Playback with noanswer option doesn't play prompt when first line in extension |
ASTERISK-06473: One-touch pause/unpause for agents (big efficiency gain). |
ASTERISK-06474: [patch][post 1.4] One-touch pause/unpause for agents (big efficiency gain). |
ASTERISK-06475: [patch][post 1.4] Don't change NATed address on re-invites |
ASTERISK-06476: memleak in chan_sip |
ASTERISK-06477: astmm crash |
ASTERISK-06478: ENUMLOOKUP crashes * with specific enums |
ASTERISK-06479: [patch] change 5060 to DEFAULT_SIP_PORT |
ASTERISK-06480: [patch] database record remains when a realtime IAX peer expires |
ASTERISK-06481: Asterisk adds SDP to a 183 In Progress?! |
ASTERISK-06482: crash in chan_sip using Monitor() |
ASTERISK-06483: asterisk-mib.txt has minor syntax error |
ASTERISK-06484: backport of OPTIMIZETALKER and pre-recorded user intros |
ASTERISK-06485: Hanging up during Name Recording Does not Clear Participant |
ASTERISK-06486: [patch] G.723.1b codec isn't compiled |
ASTERISK-06487: [patch][post 1.4] Re-introduce GSM codec capability support for chan_h323 |
ASTERISK-06488: [patch] Solaris compile generates warnings for missing alloca prototype in editline/readline.c and editline/np/vis.c |
ASTERISK-06489: MWI for IAX2 phones incorrect |
ASTERISK-06490: app_dial will not handle specific options at the same time |
ASTERISK-06491: [branch] Asterisk trunk [pre-1.4] and rfc2833 compliance |
ASTERISK-06492: Convert a string to lowercase/uppercase |
ASTERISK-06493: hide/show silenceSupp=off on SDP header |
ASTERISK-06494: [patch] Improve func-channel |
ASTERISK-06495: [patch] IFMODULE() - report if module is loaded or not |
ASTERISK-06496: RealTime Voicemail fails when field voicemail_users:review is set to yes |
ASTERISK-06497: "set debug" and iax2 calls - flooding of socket_read lines |
ASTERISK-06498: Asterisk is not constructing the BYE request RURI correctly from contact headers in some situations. |
ASTERISK-06499: MeetMe 'i' flag does not play hasleft/hasjoin sounds |
ASTERISK-06500: jitterbuffer:core dump. |
ASTERISK-06501: Asterisk coredumps |
ASTERISK-06502: Dialplan Function REGEX not behaving correctly |
ASTERISK-06503: [patch] type error in UPGRADE.txt |
ASTERISK-06504: [patch] Realtime queue members have been broken |
ASTERISK-06505: [patch][post 1.4] Random Periodic Announcements in app_queue |
ASTERISK-06506: [patch] new manager action: SendDTMF |
ASTERISK-06507: [patch] Fix for CallerID on Indian PSTN |
ASTERISK-06508: Asterisk 1.2.5 crashed when using call queues. |
ASTERISK-06509: four unnecessary lines of code |
ASTERISK-06510: Cannot detect the DTMF and FSK/ETSI before first ring |
ASTERISK-06511: MixMonitor stops recording after a random period |
ASTERISK-06512: SIP Callers to Queue Get Dumped |
ASTERISK-06513: **Unknown** channel preceeds crash |
ASTERISK-06514: duplicate in-band call progress information (duplicate ring tone) |
ASTERISK-06515: Monitor()'ed calls end up out of sync |
ASTERISK-06516: Setting up Directed call support |
ASTERISK-06517: Mixmonitor stops recording call after random time |
ASTERISK-06518: Accountcode of second leg in cdrs on attended transfer |
ASTERISK-06519: Upgrade to 2.6.9-34.ELsmp and zaptel can not compile |
ASTERISK-06520: Message playback order |
ASTERISK-06521: [patch] Fully initialize NULL frames |
ASTERISK-06522: ENUMLOOKUP is not looking up :) |
ASTERISK-06523: [patch] Voicemail reuses variables in an "interesting" manner |
ASTERISK-06524: [patch] [post-1.4] Empty strings in database is not returned as a variable value |
ASTERISK-06525: dial option 'p' doesn't work with queues. |
ASTERISK-06526: Asterisk does not compile on Linux/parisc because k6 optimiations are a target |
ASTERISK-06527: [patch] Add options to disable overlap dialling and subscriptions |
ASTERISK-06528: Dial in AGI script exits randomly with empty status variables |
ASTERISK-06529: Makefile in asterisk/codecs/gsm make complains about missing a endif |
ASTERISK-06530: Order of sounds make no sense in French |
ASTERISK-06531: [patch] missing endif in codecs/gsm/Makefile |
ASTERISK-06532: [patch] DateTime crashes on AMD64 system |
ASTERISK-06533: Audio file playback distorts on EM64T machine |
ASTERISK-06534: [Feature Request]: uniqueid in RDNIS |
ASTERISK-06535: I have added a time-compressed recording & catchup mode to app_meetme.c - should I submit it? |
ASTERISK-06536: Create AGI & DeadAGI hybrid |
ASTERISK-06537: [patch] Race condition? User leaves voicemail, SIP phone notifies desk user of message, user checks, message blank |
ASTERISK-06538: [patch][post 1.4] voicemail external notification not working |
ASTERISK-06539: Problems with GROUP variables on SIP transfers |
ASTERISK-06540: [patch] chan_sip -- fix -Werror building |
ASTERISK-06541: [patch] cdr_sqlite -- fix -Werror building |
ASTERISK-06542: [patch] Can't build IAX without SCHED_MULTITHREAD |
ASTERISK-06543: chan_alsa randomly crashing asterisk |
ASTERISK-06544: [patch] rawplayer consumes all CPU |
ASTERISK-06545: [patch] app_rpt: use ast_string_field_set instead of strncpy |
ASTERISK-06546: When no, or improper CID number is specified by client, CALLERID(num) is set to all numbers within the [context] in sip.conf |
ASTERISK-06547: Reinvite failures |
ASTERISK-06548: [patch] incapsulation of ast_best_codec(chan->nativeformats) |
ASTERISK-06549: [PATCH] Show Agents incorrectly states that agents are available while wrapping up |
ASTERISK-06550: [patch] ast_get_readformat(chan) added |
ASTERISK-06551: On SIP subscribe for the call completion function Asterisk returns a "489 Bad Event" |
ASTERISK-06552: [patch]] flash zap trunk from softphone or IP handsets |
ASTERISK-06553: When using realtime configuration of queues, queue configuration does reflect changes in the database |
ASTERISK-06554: [patch] app_meetme Muting and Manager API enhancements |
ASTERISK-06555: [patch] Bug and bugfix for ast_frisolate |
ASTERISK-06556: Zap channel may use wrong cidsignalling in ss_thread |
ASTERISK-06557: [patch ] Release the media-path when DTMF is sent outside of the media stream (SIP re-invites) |
ASTERISK-06558: qualify not working with UIP200 phones unless nat=never is in the global section of sip.conf |
ASTERISK-06559: [patch] Compile warnings from ast_expr2.y on MacOS X |
ASTERISK-06560: [patch] Allow MeetMe to use ',' deliminators in meetme.conf |
ASTERISK-06561: Asterisk duplicates results for enumlookups |
ASTERISK-06562: On SIP subscribe for the call completion function Asterisk returns a "489 Bad Event" (#6728) |
ASTERISK-06563: MacroIf causes blind transfer to fail |
ASTERISK-06564: [patch] Add registrar dialstring in realtime |
ASTERISK-06565: member_config is returning nothing during reload_queue_rt |
ASTERISK-06566: [patch] addition to the Makefile of asterisk-addons (make samples, echo...) |
ASTERISK-06567: GotoIfTime function does not handle the [weekend] parameter properly |
ASTERISK-06568: leak on ast_channel_free |
ASTERISK-06569: [patch] features.conf.sample update |
ASTERISK-06570: iax - no audio |
ASTERISK-06571: Zap channels randomly drop |
ASTERISK-06572: [patch] do not delete a call-file whose timestamp has not expired |
ASTERISK-06573: small fix to be able to set independent PEER channel variables in Dial at answer time |
ASTERISK-06574: [patch] attended transfer use transferer context first and set who is transfering at the beginning |
ASTERISK-06575: Allow modifying channel with effect |
ASTERISK-06576: [branch] SQLite 3 support for asterisk CDR |
ASTERISK-06577: Asterisk-sounds has duplicate phonetic sounds. |
ASTERISK-06578: Changing the event flag of Newexten AMI event. |
ASTERISK-06579: Tones for Malaysia |
ASTERISK-06580: generate a log message when a Required: header is received |
ASTERISK-06581: [patch] Whitespace fixes (and others) for chan_skinny |
ASTERISK-06582: The possibility RTP IP address to be different from SIP Server |
ASTERISK-06583: Need of records in CDR for Attended Transfer |
ASTERISK-06584: [patch] configurable timeout for Attended Transfer |
ASTERISK-06585: Return BUSY signal when other party is busy at Attended Transfer |
ASTERISK-06586: In sample.call missing description of "Account" parameter |
ASTERISK-06587: "usereqphone" doesn't work |
ASTERISK-06588: "type=user" doesn't work |
ASTERISK-06589: "stale nonce" with hardware voip gate |
ASTERISK-06590: Getting loads of "acl.c: Testing...." in log file. Why? |
ASTERISK-06591: Reload/Restart clears sip peer information with Realtime |
ASTERISK-06592: [patch] Asterisk with chan_skinny crash with Cisco 7920 |
ASTERISK-06593: two identical code blocks in chan_sip.c in function process_sdp |
ASTERISK-06594: [patch] does not goto context in queues.conf before second announcement |
ASTERISK-06595: callerid= in zapata.conf not used for calls coming in on that PRI |
ASTERISK-06596: Polycom+Snom cannot answer calls when sip.conf [general] callerid="Seemingly Legit" <5551212> |
ASTERISK-06597: [patch][post 1.4] FastAGI does not pass paramaters to the FastAGI Server |
ASTERISK-06598: [patch] ast_pbx_outgoing_cdr_failed description fix |
ASTERISK-06599: Verbose levels not enforced in app_voicemail |
ASTERISK-06600: Unnecessary Unlink and Link Events generated and send when digit pressed on phone |
ASTERISK-06601: [patch] [needs testing] app_voicemail looks for VMBox in the [default] context only |
ASTERISK-06602: Asterisk crashes when res_features receives a bogus frame |
ASTERISK-06603: Losing inbound caller ID when call is transfered from one sip phone to another. |
ASTERISK-06604: ReadFile causes Asterisk to crash when non-existing file is provided |
ASTERISK-06605: [Patch] added parsing for festivalcommand variable from config file |
ASTERISK-06606: DTMF tones leaking from person on hold on IAXy |
ASTERISK-06607: [patch] [post 1.4] Add option to support regexten addition and removal on qualify status |
ASTERISK-06608: keyword ringcadance used in indications.conf is misspelt |
ASTERISK-06609: [patch] Fixes trivial bug in image.c |
ASTERISK-06610: Disconnecting while meetme is palying greet message causes Asterisk to freeze whole machine |
ASTERISK-06611: [branch] txgain/rxgain R/W support on func_channel |
ASTERISK-06612: When page for more than 40 or 50 users asterisk crash |
ASTERISK-06613: [patch] new command: show profile (and update to show threads) |
ASTERISK-06614: Queues with realtime causes persistentmembers to not function |
ASTERISK-06615: group= variable in zapata.conf not initialized |
ASTERISK-06616: Getting ring back modulation on 79XX phones. |
ASTERISK-06617: voicemail access crashes server |
ASTERISK-06618: process forked by MoH is not killed on asterisk shutdown |
ASTERISK-06619: chan_sip undefined symbol |
ASTERISK-06620: [patch] S_OR janitor |
ASTERISK-06621: [patch] sendonly RTP stream handling |
ASTERISK-06622: [patch] junitor project -- S_OR |
ASTERISK-06623: crash / segfault (or other evil) when call is ended while MixMonitoring |
ASTERISK-06624: volume on record ('r' option) are 12dbi too loud |
ASTERISK-06625: IAX2 native transfer no audio with jitterbuffer=yes |
ASTERISK-06626: Unable to hear voice |
ASTERISK-06627: for loops execute iteration (arg 3) after initialization (arg 1) |
ASTERISK-06628: [patch] Add manager events to report meetme mute/unmute actions |
ASTERISK-06629: [branch] SSL encryption for Asterisk Manager Interface (AMI) |
ASTERISK-06630: [quick fix] Typo in acl.c, from recent tos stuff |
ASTERISK-06631: CLOBBERMMX warning after defining CONFIG_ZAPTEL_MMX |
ASTERISK-06632: missing quotes in contrib/init.d/rc.redhat.asterisk |
ASTERISK-06633: [patch] Allow to assign names to nethdlc interfaces. |
ASTERISK-06634: [patch] S_OR janitor -- last patch |
ASTERISK-06635: [patch] IAX2 send_packet small cleanup/optimize |
ASTERISK-06636: RealTime and codecs order |
ASTERISK-06637: SIP channels hang in semi-active state |
ASTERISK-06638: [patch] devicestate problem |
ASTERISK-06639: Join/Leave (i) option, notifies the leaves with incorrect name. |
ASTERISK-06640: Registration time out after few minutes |
ASTERISK-06641: meetme - no audio for new user during join announcment |
ASTERISK-06642: Missing default 'default' case for switches |
ASTERISK-06643: Operation of System (used to call shell script) has changed in recent version of Asterisk |
ASTERISK-06644: [patch] optionally record audio of Page command, to then make a ReplayLastPage sound available |
ASTERISK-06645: Problem with cdr_odbc |
ASTERISK-06646: [patch] asprintf missing on Solaris Blocks compiliation |
ASTERISK-06647: GosubIf cmd doesn't handle regular expressions like GotoIf, the result is always FALSE |
ASTERISK-06648: [patch] [need disclaimers] retransmission-related segfault in chan_sip |
ASTERISK-06649: [patch] IAX2 send_packet fix using pointer before checking that it not NULL |
ASTERISK-06650: [patch] chan_iax2.c -- small cleanups |
ASTERISK-06651: [patch] code clean in chan_zap.c |
ASTERISK-06652: [patch] more clear text in gotoif |
ASTERISK-06653: [patch] minor change and code clean in app_stack.c |
ASTERISK-06654: [patch] can't build RPM in RedHat EL 4 |
ASTERISK-06655: [patch] Signal that call duration limit is enforced + whitespace cleanup |
ASTERISK-06656: [patch] typo in extensions.conf.sample |
ASTERISK-06657: current revisions of chan_iax2 seem to be unable to complete calls with revisions of chan_iax2 on 1.2.X systems. |
ASTERISK-06658: [PATCH] Allow Caller Name from CPE to Master on ISDN NI1 |
ASTERISK-06659: res_musiconhold.so crashes on startup if pointed at directory w/ too many files |
ASTERISK-06660: [branch] add a parameter to app_queue to allow for an AGI/FastAGI script to be executed |
ASTERISK-06661: ChanIsAvail always uses priority juming even if the option is not set explicitly |
ASTERISK-06662: Call Pickup with *8 does not check callgroup & pickupgroup settings (SIP) |
ASTERISK-06663: Logger.c Cause segfault |
ASTERISK-06664: Crash on starting asterisk |
ASTERISK-06665: Asterisk does not process 'message-summary' subscriptions. |
ASTERISK-06666: Possible bug in size allocations |
ASTERISK-06667: GSM fails to build on amd64 systems |
ASTERISK-06668: [patch] billsec and duration record only max int (x86_64) |
ASTERISK-06669: [patch] Remove duplicated break, fix ident/formatting, update cli help |
ASTERISK-06670: [patch] Fix for 2 minor AMI issues with chan_sip.c |
ASTERISK-06671: New [+context] feature in extensions.conf allowing more stuff to be added to an existing context as the dialplan loads |
ASTERISK-06672: Give a more complete error message |
ASTERISK-06673: Dial(type/identifier, timeout, m(message)) doesn't send RTP till some RTP arrives |
ASTERISK-06674: [patch] Silent verboser, "unify" constructs, move some code from inside "if (option_verbose > x)" constructs |
ASTERISK-06675: [patch] IAX2 simplification |
ASTERISK-06676: [branch] chan_skinny-fixup |
ASTERISK-06677: IP fones LOCKUP 3-4 times a day |
ASTERISK-06678: [patch] make astman_append honor writetimeout |
ASTERISK-06679: [patch] no need to init sockaddr_in struct twice |
ASTERISK-06680: Monitor option b causes 1 second audio delay |
ASTERISK-06681: [patch] System freeze: responds to ping, accepts connections on ports (5060, 80, etc.) but does not respond back |
ASTERISK-06682: Call-limit does not work on realtime |
ASTERISK-06683: [patch] No Manager event on loss/return of E1 |
ASTERISK-06684: AEL & macros |
ASTERISK-06685: Peer registration timeout doesn't send manager event |
ASTERISK-06686: [patch] Allow contexts in regexten so that extensions can be added to multiple contexts when peer registers |
ASTERISK-06687: [feature request] Cannot specify *70 for parkext or *71-*79 for parkpos |
ASTERISK-06688: 180 RINGING Message immediately after INVITE - Causing problems at sipendpoints handling 480 Busy here message. |
ASTERISK-06689: SIP: DTMF INFO before answer causes disconnect |
ASTERISK-06690: [patch] Poor use of rand() scaling |
ASTERISK-06691: PATCH: DBDel for the Manager Interface |
ASTERISK-06692: Maximum pbx stack exceeded |
ASTERISK-06693: threewaycalling=yes may cause unexpected "transfers" with analog phones |
ASTERISK-06694: Hanging up error |
ASTERISK-06695: [patch] manager_event DNDState is inside 'if (option_verbose > 2)' block |
ASTERISK-06696: asterisk crash in chan_iax2 |
ASTERISK-06697: [patch] continue is inside 'if (option_verbose >= 2)' block |
ASTERISK-06698: Can't get Pickup app working |
ASTERISK-06699: [patch] move 'res = -1' outside if (option_verbose... block |
ASTERISK-06700: [patch] Silent some compiler warnings |
ASTERISK-06701: [patch] formatting + spacing |
ASTERISK-06702: ast_channel_spy_read_frame doesn't return a value |
ASTERISK-06703: [patch] app_exec always returns 0 and sets channel-variable RESULT instead |
ASTERISK-06704: Directory lookup fails with no voicemail password |
ASTERISK-06705: [patch] Silent verbose and debug output when not enabled |
ASTERISK-06706: [patch][post-1.4] Convert ast_verbose into macros? |
ASTERISK-06707: SIP 100 Trying not being sent on 404 error |
ASTERISK-06708: [patch] res/Makefile broken after 17628 commit |
ASTERISK-06709: [patch] main Makefile cleanup after 17735 |
ASTERISK-06710: [patch] compiler warnings on res_config_pgsql |
ASTERISK-06711: Asterisk crash II |
ASTERISK-06712: Bad chars in incoming callerid my cause cdr_csv records with illegal format |
ASTERISK-06713: wav49 format is garbled in latest SVN trunk. |
ASTERISK-06714: Features beginning with '*' dont work for chan_agent |
ASTERISK-06715: Not all voicemail message formats deleted when message deleted in latest SVN trunk. |
ASTERISK-06716: [patch] Remove OSP support code from SIP channel to res_osp.c and app_osplookup.c |
ASTERISK-06717: O'Clock (heure) said twice in French |
ASTERISK-06718: [asterisk-addons] Milliwatt analyzer |
ASTERISK-06719: Park ignores the extension argument passed to it |
ASTERISK-06720: [patch] ast_play_and_record() with silence detection doesn't trim the silence |
ASTERISK-06721: [patch] Remove look.c entries in apps/Makefile |
ASTERISK-06722: [patch] No need to link asterisk binary with ncurses |
ASTERISK-06723: [patch] Added application MacroIfTime |
ASTERISK-06724: [patch] app.c cleanup |
ASTERISK-06725: [patch] SMDI crash |
ASTERISK-06726: Chanspy does not output one-way audio |
ASTERISK-06727: [branch][post 1.4] play agent num to caller in queue |
ASTERISK-06728: [patch] Empty recordings with length of 4 seconds |
ASTERISK-06729: [patch] Disable debug output when option_debug not set. |
ASTERISK-06730: Possible Integer overflow |
ASTERISK-06731: incorrect msgNNNN.txt.tmp files left behind when operator=yes and hitting 0 while leaving a message |
ASTERISK-06732: [patch] channels/Makefile cleanup |
ASTERISK-06733: Anti-Ex-Girlfriend not working on RealTime Extensions - 1.2.6 |
ASTERISK-06734: [patch] Junitor -- some patches for res_pgsql.c |
ASTERISK-06735: Asterisk randomly segfaults - Appears to be chan_iax2 |
ASTERISK-06736: Need to add a better diagnostic message when mkstemp() fails |
ASTERISK-06737: [patch] rev 18607 commit breaks coding guidelines |
ASTERISK-06738: [patch] Add 'make config' for debian distro |
ASTERISK-06739: [no patch] r18666 breakage of chan_sip (peer cli completion) |
ASTERISK-06740: play agent num to caller in queue when the agent picked up the headset |
ASTERISK-06741: IAX dial string not working in queues.conf |
ASTERISK-06742: [patch] http show status always show usage info |
ASTERISK-06743: [patch] convert http and http_manager to res_* |
ASTERISK-06744: On SIP REGISTER event trigger |
ASTERISK-06745: [patch] func_odbc -Werror building fix |
ASTERISK-06746: codec_g729a.so loading problems |
ASTERISK-06747: Added new option 'n' to Page() app to keep the Page from including the caller. |
ASTERISK-06748: Festival.conf specifying cachedir |
ASTERISK-06749: [patch] outgoing RTP stream has a big timestamp gap after Asterisk recieves a reinvite on the other leg |
ASTERISK-06750: [patch] do not allow unlimited calls if app_dial is called with S(0) or L(0), or if there is an error in parsing limit values |
ASTERISK-06751: segfault in malloc and calloc |
ASTERISK-06752: [post 1.4][patch] move ast_carefulwrite() to utils.h/.c because it is used by cli.c in ast_cli() |
ASTERISK-06753: Read Extension Number After Name |
ASTERISK-06754: i getting this error with asterisk |
ASTERISK-06755: asterisk could really use a suite of tests and test scripts |
ASTERISK-06756: chan_local support for L and S flag on app_dial won't work at all |
ASTERISK-06757: [patch] get rid of cli.c recompilation every time asterisk/version.h is updated |
ASTERISK-06758: Originate records Disposition 'answered' in cdr when 'originating device' answers |
ASTERISK-06759: AGI behavior on fail |
ASTERISK-06760: [patch][post 1.4] app_voicechanger.c |
ASTERISK-06761: media path not always opened when needed |
ASTERISK-06762: ODBC crashes whenever I issue any valid ODBC-related command... |
ASTERISK-06763: Voicemail broadcast to large number of extensions fails after 256 characters |
ASTERISK-06764: [patch] asterisk does not build when there is static ncurses library present on a box |
ASTERISK-06765: AgentMonitorOutgoing channel.c:787 channel_find_locked |
ASTERISK-06766: app_queue state change failing - loop/deadlock |
ASTERISK-06767: [patch] show last registration time in 'sip show registry' |
ASTERISK-06768: Segfault on show channels after Agent is removed from Meetme |
ASTERISK-06769: [patch][post 1.4] Change default return extension after parking timeout |
ASTERISK-06770: Crash on Dial(xxx,xxx,m(xxx)) when 2nd call arrives |
ASTERISK-06771: zap restart without asterisk restart |
ASTERISK-06772: [patch] Skip recording, just play a message |
ASTERISK-06773: extensions.conf parser cannot handle certain global variable names |
ASTERISK-06774: channel.c randomly producing "Avoided Initial Deadlock ... 10 retries!" message |
ASTERISK-06775: channel.c randomly producing "Avoided Initial Deadlock ... 10 retries!" message |
ASTERISK-06776: Libpri doest set pricause |
ASTERISK-06777: [patch] Allow the attachment format to be specified differently for different mailboxes |
ASTERISK-06778: More Functionility Required in RetryDial Application |
ASTERISK-06779: [patch] pri_dchannel: Unable to move channel |
ASTERISK-06780: Patch: Voice mail in Polish support |
ASTERISK-06781: [patch] zaptel fails to make devices on not-udev systems |
ASTERISK-06782: [patch] No need to call ast_best_codec and ast_codec_choose when there is disallow=all in [general] |
ASTERISK-06783: [patch] data dir for read-only data |
ASTERISK-06784: loader changes feedback |
ASTERISK-06785: Channel.h (etc) supports sending text, html, and images, but not receiving them |
ASTERISK-06786: [patch] Voice mail in Polish support |
ASTERISK-06787: [patch] Thailand tonezone data |
ASTERISK-06788: MYSQL app crashes after reload |
ASTERISK-06789: attended transfer feature - calls getting lost |
ASTERISK-06790: FindMe/FollowMe App not acknowledging: press 1 to connect the caller |
ASTERISK-06791: [patch] Junitor -- use new ast_channel_(lock|unlock|trylock) in channel.c |
ASTERISK-06792: Intermittent crash |
ASTERISK-06793: Problem with '183 Session Progress' |
ASTERISK-06794: [patch] format_mp3 update |
ASTERISK-06795: [patch] L option of Dial does not work properly (new case) |
ASTERISK-06796: [patch] enable multiple voicesets for other languages than 'en' |
ASTERISK-06797: [patch] Move res_osp.c into app_osplookup.c for the new way of loading |
ASTERISK-06798: [patch] load_realtime_queue with dbconfig overrides dynamic configuration |
ASTERISK-06799: Sip Registration Stale Nonce Problem on long-latency links (satellite) |
ASTERISK-06800: application random() is never true when probability is 1 (as in random(1:$ARGS)) |
ASTERISK-06801: Incoming SIP calls that do not provide auth are lumped into one BIN and screws up data and feature interaction |
ASTERISK-06802: [patch] wrong event structure used in handling of PRI_EVENT_SETUP_ACK |
ASTERISK-06803: Zaptel installation |
ASTERISK-06804: Function REGEX gets confused when using curly braces in a regular expression |
ASTERISK-06805: [patch] app_voicemail fails to compile with ODBC_STORAGE |
ASTERISK-06806: Manager Event: DTMF |
ASTERISK-06807: [patch] Two memory leaks in ast_play_and_prepend -> app.c rewrite |
ASTERISK-06808: [patch][post 1.4] EndDial Manager Event |
ASTERISK-06809: DTMF and IAX and transcoding creates extra codec translation instances |
ASTERISK-06810: nested while loops dont return to parent loop when you goto() out of the child loop |
ASTERISK-06811: Asterisk has problems working if dns name resolves to several IP |
ASTERISK-06812: SIP_HEADER documentation |
ASTERISK-06813: An attempt to transfer call into same callid causes deadlock |
ASTERISK-06814: [branch][post 1.4] AAL -- Asterisk Argument Language |
ASTERISK-06815: test_this_branch fails to compile |
ASTERISK-06816: alphanumeric username in register to Sip-Proxy (sip.conf) makes [/extension] not working |
ASTERISK-06817: [patch] Voicemail crashes Asterisk if user chooses to prepend when forwarding. |
ASTERISK-06818: Dial on local channel changes presentation bit |
ASTERISK-06819: SIP routing gets confused |
ASTERISK-06820: MixMonitor() causes Asterisk to deadlock under good call volume |
ASTERISK-06821: Asterisk cannot work with two different codec types in dynamic codecs negotiaation like iLBC |
ASTERISK-06822: [patch] chan_h323 does not build |
ASTERISK-06823: No more Wait/Playback/Dial in Hangup extension after upgrade to 1.2.7 |
ASTERISK-06824: [patch] Incorrect log statement when playing transfer sounds |
ASTERISK-06825: [patch] Timewaste on count messages in app_voicemail with odbc support |
ASTERISK-06826: [patch] Send voicemail from v-mail using Directory and Realtime fails |
ASTERISK-06827: libpri locks up if an external resource-hungry process is run on same system |
ASTERISK-06828: [patch] Zonedata.c file addition for tones of Pakistan |
ASTERISK-06829: QueueStatusComplete not being returned properly |
ASTERISK-06830: Stop now |
ASTERISK-06831: Stop now Crash |
ASTERISK-06832: Nonce blanked |
ASTERISK-06833: [patch] Add a function to left/right justify text |
ASTERISK-06834: Hints via RealTime |
ASTERISK-06835: Crash/hang on uclibc based systems with call files |
ASTERISK-06836: [patch] Improve CPU-performance of MG2 ec |
ASTERISK-06837: [patch] func_iconv - convert string from in-charset to out-charset |
ASTERISK-06838: res_bonjour |
ASTERISK-06839: [patch] Reloading app_queue.so unpauses all members |
ASTERISK-06840: Do not require zaptel by default |
ASTERISK-06841: [patch] Fix output delimiters and add prefix parameter |
ASTERISK-06842: Unreachable IAX peer causes silence on calls using other IAX channel |
ASTERISK-06843: [patch] List macro janitor for voicemail |
ASTERISK-06844: 'continue' in a for-loop does not execute post-iteration step |
ASTERISK-06845: Asterisk crashes unless built with 'make dont-optimize' w/ gcc 4.x on FC4/5 |
ASTERISK-06846: [patch] fix autoconf detection for chan_h323 |
ASTERISK-06847: Linksys/Sipura 941 does not ring on inbound ZAP call |
ASTERISK-06848: [patch] Fix smdi.txt's use of the depricated ${CALLERIDNUM} |
ASTERISK-06849: [post 1.4] SIP Notify with MWI re-sent for all sip users on asterisk restart or reload |
ASTERISK-06850: [patch] Asterisk 1.2.7.1 dying in alawtolin_framein at codec_alaw.c |
ASTERISK-06851: IAX2 peer (1.0.9) changing refresh value for incoming iax2 regreq from 60 to 0 (client 1.2.6/7) |
ASTERISK-06852: Dialplan function to count number of callers waiting in a queue |
ASTERISK-06853: Announce user without review |
ASTERISK-06854: [patch] Holdtime and talktime not logged in queue_log when blind transfers are made by an agent |
ASTERISK-06855: [patch] junitor project -- channel locks in rtp.c |
ASTERISK-06856: unexpected DTMF |
ASTERISK-06857: [patch] Leak in chan_ooh323 during call setup |
ASTERISK-06858: unresponsive DNS servers cause asterisk to pause for extended periods during startup |
ASTERISK-06859: Asterisk does not build on Fedora Core 5 due to defining of malloc to rpl_malloc (and rpl_realloc) |
ASTERISK-06860: [patch] Queue(somequeue,,,,) -> interpreted as Queue(somequeue,,,,0) |
ASTERISK-06861: [patch] do not access and do not try to free unallocated memory in http.c |
ASTERISK-06862: [patch] callgroup/pickupgroup output via AMI is broken |
ASTERISK-06863: [patch] allow Asterisk to set high ToS bits as non-root on Linux |
ASTERISK-06864: [feature] support for group number > 63 in channel.c |
ASTERISK-06865: Additional configure checks for chan_phone |
ASTERISK-06866: Multiple bind addresses with priority |
ASTERISK-06867: [patch] externip replacement is made regardless of 'true' source IP |
ASTERISK-06868: Asterisk causes the whole system to hang |
ASTERISK-06869: [patch] blind transfering a sip call to parking causes chan_sip to hang. |
ASTERISK-06870: [patch] aelparse linking |
ASTERISK-06871: Using FastAGI incoming trunk behavior does not match outgoing behavior |
ASTERISK-06872: [branch] define paths in a more autoconf-friendly way |
ASTERISK-06873: [patch] Junitor -- use new ast_channel_(lock|unlock|trylock) in many places |
ASTERISK-06874: [patch] fix building app_rpt.c |
ASTERISK-06875: Unable to install asterisk-addons |
ASTERISK-06876: If a user press 0 during recording, the message is not completely erased and stay stuck in the mailbox |
ASTERISK-06877: AEL commands in CLI still "ael2 ..." instead of "ael ..." |
ASTERISK-06878: [branch][post 1.4] Allow caller to dial 1-9 while leaving voicemail |
ASTERISK-06879: [patch] 4 bugs in voicemail |
ASTERISK-06880: builds of trunk fail to complete do to pbx_ael |
ASTERISK-06881: The caller can't hear ringing when jitterbuffer is enabled |
ASTERISK-06882: [patch] Audio streams may be closed twice, crashing Asterisk |
ASTERISK-06883: Asterisk crashes when using record_file to record in h263 format |
ASTERISK-06884: [patch[ some little G.711 optimizations |
ASTERISK-06885: [patch] "asterisk -x" does not disconnect session when built with "dont-optimize" |
ASTERISK-06886: [patch] Allow channels in ChanSpy to belong to mutiple spygroups |
ASTERISK-06887: Compilation error |
ASTERISK-06888: Problem with long contact lines |
ASTERISK-06889: [patch] invalid OLC/faststart with bindaddr=0.0.0.0 |
ASTERISK-06890: [patch] Only first digit of RFC2833 DTMF being sent |
ASTERISK-06891: [patch] SayNumber, Saydate, Saytime Hungarian syntax |
ASTERISK-06892: [patch] SPRINTF dialplan function |
ASTERISK-06893: Asterisk sends CANCEL on INVITE before receiving any provisional response, thus not killing INVITE |
ASTERISK-06894: chan_local inserts a "," in channel name, therefore ${CHANNEL} with verboser app results in a warning |
ASTERISK-06895: SIP answer 480 is translated incorrectly by hangup_sip2cause |
ASTERISK-06896: loss of memory in due corse of time |
ASTERISK-06897: [patch] fix for unnecessary dial delay if 2 or more early match patterns (i.e. with !) are present |
ASTERISK-06898: [patch] integrate MixMonitor with app_queue |
ASTERISK-06899: func_iconv |
ASTERISK-06900: [patch] ExecIf() should use pbx_checkcondition instead of ast_true |
ASTERISK-06901: Asterisk getting kill |
ASTERISK-06902: A proposal to introduce a "LINEAR" queue strategy |
ASTERISK-06903: Asterisk CLI Tab completion no longer works as expected |
ASTERISK-06904: Crash or block on Park with applicationmap |
ASTERISK-06905: simultaneous invalid MeetMe pins + simultaneous hangups stops Asterisk |
ASTERISK-06906: [patch] pedantic mode broken |
ASTERISK-06907: [patch] make config doesn't work other than RH distribution |
ASTERISK-06908: [patch] ExecIf is in app_while - not anymore! |
ASTERISK-06909: Channel variables do not get created when forwarding a call as a result of 302 "Moved Temporarily" response |
ASTERISK-06910: [patch] Allow the CLI to list all online agents. |
ASTERISK-06911: Hangup event not occured when both the ends are hungup |
ASTERISK-06912: incorrect PID display for asterisk -r for forked asterisk |
ASTERISK-06913: [patch] always assume pid_t is long |
ASTERISK-06914: SIP<->IAX2 call using H.263 crashes Asterisk unless built with 'make dont-optimize' |
ASTERISK-06915: Australian Indications |
ASTERISK-06916: Requests with missing From crash pedantic mode |
ASTERISK-06917: [patch] Selectively disable EC for incoming DID's from PRI |
ASTERISK-06918: [patch] Sip Require headers |
ASTERISK-06919: [patch] Macro does not fully execute when calling party disconnects |
ASTERISK-06920: [patch] no Pickup on NoCDR and unused Flag AST_CDR_FLAG_POST_DISABLED |
ASTERISK-06921: [patch] some little G.711 optimizations |
ASTERISK-06922: E1 PRI Disconnect audio message indication |
ASTERISK-06923: [patch] minor annoyance fix for app_amd config parser vs amd.conf |
ASTERISK-06924: If you use coma in your Dial String instead of "/" the data will be incorrectly entered in the table. |
ASTERISK-06925: [patch] no need in -fomit-frame-pointer |
ASTERISK-06926: [patch][post 1.4] Remove chan->language from ast_stream_and_wait call |
ASTERISK-06927: [patch] make rpm is broken due to outdated .spec file |
ASTERISK-06928: [patch] SIPGetHeader does not accept the 'j' flag (can't stop priority jumping) |
ASTERISK-06929: chan_local makes cdr 'duration' and 'billsec' useless |
ASTERISK-06930: Wrong event reported for Originate. Reports the opposite of what the event is. |
ASTERISK-06931: q931_dumpie calls pri_message multiple times |
ASTERISK-06932: Manager is not returning events properly |
ASTERISK-06933: memory leak in find_user_realtime |
ASTERISK-06934: [patch] janitor project -- if( ast_mutex_lock ) |
ASTERISK-06935: [patch] janitor project -- ast_channel_lock instead of ast_mutex_lock |
ASTERISK-06936: [patch] ability to detect unreachable or not working fastagi server |
ASTERISK-06937: subscriptions do not work if sipdomains are used |
ASTERISK-06938: [patch] callingpres ignored when callerid not set |
ASTERISK-06939: Asterisk fails calls that contain multipart SDP |
ASTERISK-06940: [patch] .txt file stranded if user deletes message while a message is being left |
ASTERISK-06941: [patch] If a queue has only a paused member i get "No one is answering queue" |
ASTERISK-06942: [patch] Junitor -- use new ast_channel_(lock|unlock|trylock) in res_monitor.c |
ASTERISK-06943: queue timeout with penalty avoid switching between agents |
ASTERISK-06944: Asterisk do not validate SDP Session Name "Inacive" |
ASTERISK-06945: [patch] MeetMeAdmin Additional Features (Volume) |
ASTERISK-06946: [patch] When selecting multiple columns commas are converted to pipes resulting in query failure. |
ASTERISK-06947: [patch] When using ODBC storage message count always results in error |
ASTERISK-06948: [patch] When using ODBC storage, app_voicemail has a file descriptor leak |
ASTERISK-06949: Set TIMEOUT(absolute) breaks DIAL timeout parameter when dialing from SIP to SIP |
ASTERISK-06950: Accountcode isn't accurate with ForkCDR. |
ASTERISK-06951: [patch] Dependency system is borken |
ASTERISK-06952: [patch] aelparse will not build with astmm enabled |
ASTERISK-06953: Endless loop message bug |
ASTERISK-06954: [patch] OSPNext does not handle success/failure correctly |
ASTERISK-06955: [patch] smsq build on x86_64 with -Werror |
ASTERISK-06956: new dialplan prioritizaiton loses "match by Caller*ID" |
ASTERISK-06957: chan_iax2.c does not see 'tos' variable and always reports depricated use |
ASTERISK-06958: Performing 'extensions reload' clears all Regexten NoOps |
ASTERISK-06959: [branch] Ability to set HANGUPCAUSE sent by Hangup() to channel manually |
ASTERISK-06960: [patch] cdr_pgsql -- using unix sockets if host not set |
ASTERISK-06961: [patch] * does not compile |
ASTERISK-06962: MixMonitor stop recording when bridge on Asterisk 1.2.7.1 |
ASTERISK-06963: [branch] New custom SQLite3 CDR driver backend (cdr_sqlite3_custom) |
ASTERISK-06964: 'make install' overwrites modprobe.conf on FC3 |
ASTERISK-06965: [patch] Add support to search for first OR last name in the directory |
ASTERISK-06966: Changing codec dynamically |
ASTERISK-06967: [patch] cdr_pgsql -- some cleanups |
ASTERISK-06968: Changing codec dynamically |
ASTERISK-06969: Provide a snmp application to monitor digium cards |
ASTERISK-06970: Asterisk crash in sip_alloc at chan_sip.c:3038 |
ASTERISK-06971: addons are compiled with errors |
ASTERISK-06972: addons are compiled with errors |
ASTERISK-06973: [patch] chan_zap does not read the minidle parameter from zapata.conf |
ASTERISK-06974: Crash in channel.c:queue_frame_to_spies() |
ASTERISK-06975: No reply to PUBLISH requests |
ASTERISK-06976: [patch] IAX2 MWI fix (# of messages incorrectly reported) |
ASTERISK-06977: Priority jumping not working on VoiceMail app with new sintax |
ASTERISK-06978: SIP call blind-transfered into Parking makes it stop working |
ASTERISK-06979: [patch] can't set AMA flags |
ASTERISK-06980: VMCOUNT() does not work with ODBC_STORAGE. |
ASTERISK-06981: [patch] don't stop recording until hangup (no terminator) |
ASTERISK-06982: [patch] Change group of voicemails for vmail.cgi security |
ASTERISK-06983: voicemail to more than one mailbox fails |
ASTERISK-06984: func_odbc write values missing |
ASTERISK-06985: Implementing Paging on the Linksys SPA9XX phones |
ASTERISK-06986: Asterisk crashes in chan_h323 with a loop in dialpan |
ASTERISK-06987: enumlookup function bug |
ASTERISK-06988: Crash/coredump in expire_register in jitterbuffer-1.2 |
ASTERISK-06989: Polycom 601 SP 500 Internal Server Error in response to SIP Notify messages |
ASTERISK-06990: Userevent command and AEL, wrong parsing ? |
ASTERISK-06991: DTMF Tones ("talkback") on all kind of Channels |
ASTERISK-06992: Spurious check for qe->parent->autofill in app_queue |
ASTERISK-06993: Malicious (or dumb) user forwarding their own extension to their phone. |
ASTERISK-06994: monitor_format under realtime (mysql) causes crash on bridging agent to caller |
ASTERISK-06995: pressing 0 in call and waiting causes messagexxx.txt file to be left behind. |
ASTERISK-06996: We can't register behind NAT to our sip server. |
ASTERISK-06997: Asterisk segfault on bogus frame from Agent Channel |
ASTERISK-06998: Wrong account selected on inbound calls |
ASTERISK-06999: [patch] compile error with asterisk-addons in the trunk |